kernel-fxtec-pro1x/asoc/msm-pcm-q6-v2.h
Laxminath Kasam 605b42f92c audio-lnx: Rename folders to new flat structure.
Kernel audio drivers can be categorised into below folders.
asoc - ALSA based drivers,
asoc/codecs - codec drivers,
ipc - APR IPC communication drivers,
dsp - DSP low level drivers/Audio ION/ADSP Loader,
dsp/codecs - Native encoders and decoders,
soc - SoC based drivers(pinctrl/regmap/soundwire)

Restructure drivers to above folder format.
Include directories also follow above format.

Change-Id: I8fa0857baaacd47db126fb5c1f1f5ed7e886dbc0
Signed-off-by: Laxminath Kasam <lkasam@codeaurora.org>
2017-08-18 16:56:12 -06:00

130 lines
3.1 KiB
C

/*
* Copyright (C) 2008 Google, Inc.
* Copyright (C) 2008 HTC Corporation
* Copyright (c) 2012-2017 The Linux Foundation. All rights reserved.
*
* This software is licensed under the terms of the GNU General Public
* License version 2, as published by the Free Software Foundation, and
* may be copied, distributed, and modified under those terms.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
*
* See the GNU General Public License for more details.
* You should have received a copy of the GNU General Public License
* along with this program; if not, you can find it at http://www.fsf.org.
*/
#ifndef _MSM_PCM_H
#define _MSM_PCM_H
#include <dsp/apr_audio-v2.h>
#include <dsp/q6asm-v2.h>
/* Support unconventional sample rates 12000, 24000 as well */
#define USE_RATE \
(SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT)
extern int copy_count;
struct buffer {
void *data;
unsigned int size;
unsigned int used;
unsigned int addr;
};
struct buffer_rec {
void *data;
unsigned int size;
unsigned int read;
unsigned int addr;
};
struct audio_locks {
spinlock_t event_lock;
wait_queue_head_t read_wait;
wait_queue_head_t write_wait;
wait_queue_head_t eos_wait;
wait_queue_head_t enable_wait;
wait_queue_head_t flush_wait;
};
struct msm_audio_in_frame_info {
uint32_t size;
uint32_t offset;
};
#define PLAYBACK_MIN_NUM_PERIODS 2
#define PLAYBACK_MAX_NUM_PERIODS 8
#define PLAYBACK_MAX_PERIOD_SIZE 122880
#define PLAYBACK_MIN_PERIOD_SIZE 128
#define CAPTURE_MIN_NUM_PERIODS 2
#define CAPTURE_MAX_NUM_PERIODS 8
#define CAPTURE_MAX_PERIOD_SIZE 122880
#define CAPTURE_MIN_PERIOD_SIZE 320
struct msm_audio {
struct snd_pcm_substream *substream;
unsigned int pcm_size;
unsigned int pcm_count;
unsigned int pcm_irq_pos; /* IRQ position */
uint16_t source; /* Encoding source bit mask */
struct audio_client *audio_client;
uint16_t session_id;
uint32_t samp_rate;
uint32_t channel_mode;
uint32_t dsp_cnt;
int abort; /* set when error, like sample rate mismatch */
bool reset_event;
int enabled;
int close_ack;
int cmd_ack;
/*
* cmd_ack doesn't tell if paticular command has been sent so can't
* determine if it needs to wait for completion.
* Use cmd_pending instead when checking whether a command is been
* sent or not.
*/
unsigned long cmd_pending;
atomic_t start;
atomic_t stop;
atomic_t out_count;
atomic_t in_count;
atomic_t out_needed;
atomic_t eos;
int out_head;
int periods;
int mmap_flag;
atomic_t pending_buffer;
bool set_channel_map;
char channel_map[8];
int cmd_interrupt;
bool meta_data_mode;
uint32_t volume;
bool compress_enable;
/* array of frame info */
struct msm_audio_in_frame_info in_frame_info[CAPTURE_MAX_NUM_PERIODS];
};
struct output_meta_data_st {
uint32_t meta_data_length;
uint32_t frame_size;
uint32_t timestamp_lsw;
uint32_t timestamp_msw;
uint32_t reserved[12];
};
struct msm_plat_data {
int perf_mode;
struct snd_pcm *pcm;
};
#endif /*_MSM_PCM_H*/